Media Capture and Streams: WebRTC allows browsers and applications to access audio and video devices, such as microphones and cameras, and capture media streams in a standardized way.

Peer-to-Peer Communication: WebRTC establishes direct communication channels between peers, enabling them to send audio, video, and data directly to one another without intermediaries.

Real-Time Communication: WebRTC is designed for low-latency, real-time communication, making it suitable for applications where instantaneous feedback is crucial, such as video conferencing and online gaming.

Secure Communication: Security is a fundamental aspect of WebRTC. It uses encryption protocols to secure data transmission, ensuring that communications remain private and protected from eavesdropping.

NAT Traversal: WebRTC includes mechanisms for traversing Network Address Translation (NAT) devices and firewalls, making it easier to establish direct connections between peers, even in complex network environments.

STUN and TURN Servers: To assist in establishing connections, WebRTC can use STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers. These servers help identify network configurations and relay data when direct peer-to-peer connections are not possible.

WebRTC APIs: WebRTC provides JavaScript APIs that developers can use to implement real-time communication features in web applications. These APIs include getUserMedia for accessing media devices, RTCPeerConnection for setting up and managing peer-to-peer connections, and RTCDataChannel for sending arbitrary data between peers.